similar to: Dropping numbers on dialout through tdm400p

FC2, Asterisk 1.0.0, Zaptel 1.0.0 TDM400P Port 1 FXS Port 4 FXO Standard analogue handset plugged in with pstn line. Problem: I have 2 analog phones that I use, when plugged directly into pstn line both phones work perfectly, dialing no issues. When I plug the handsets into the TDM400P, one works perfectly the other drops random numbers. Its like the tone is slightly different on the second

Scenario, I got some very good help earlier from Joseph getting me up and started but I have a couple of small problems still. Setup: FC2 Asterisk 1.0 Zaptel 1.0, TMP400P, FXS Port 1 FXO Port 4 Analog dialout line and Analog handset plugged in. Problems: 1. Incoming calls work and the phone rings and can be answered no problems, (although I wouldn't mind being able to adjust the ring but

Hi, I have a weird situation where I have a noop command putting the callerid of the caller on my asterisk console so I know who is calling as a test, but it is putting the callerid of my extension in instead of the callerid of the incoming line. My /etc/asterisk/zapata.conf is [channels] context=default ;switchtype=national usecallerid=yes cidsignalling=v23 cidstart=polarity hidecallerid=no

Sorry to post such a newb set of questions but I have been hammering about trying to get Asterisk running on FC2 machine reading everything available (I think that is what stuffed me, shouldn't have read it all :-) ). Config FC2 running Asterisk 1.0.0, with the h323 compiled in and installed correctly. Amazingly enough I have everything compiled correctly and installed. I am running a

FC2 Asterisk 1.0 When I dial a H323 dialup to an existing OKI Voip Router (BV1250), I get an EndedByRefusal yet the OKI Gateway is setup with the corrent reply ip addresses etc etc, unfortunately its an existing multiple voip router setup with g723.1 and g729a, so changing the codec on the router maybe an issue. I have compile in the h323 as per the channels/h323 setup with the listed libraries.

Asterisk 1.0.7 Has anyone got the hint function working, and maybe with the GXP2000. I am testing with 2 GXP2000 phones (firmware 1.0.1.9) at the moment trying to get the LED's to light up. On ext 690, button 1 is setup for ext 691, I did this using both methods 691, and <sip:691@192.168.69.1> On ext 691, button 1 is setup for ext 690, I did this using both methods 690, and

Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing

Hi, I am sorry to be asking this but the wiki is down and has been for a couple of days and I need to get this working before Monday to get my live system setup. Trying to get the Snom 190's and soon to arrive 3com 3102's to use the function keys and for the life of me I can't work it out from the conversations on the archive what I am going exactly wrong here? The snom 190 with

Hi Everyone, Merry Christmas :-).... My Asterisk Box doesn't have a sound card, it is running Asterisk 1.02 Zaptel 1.02 Libpri 1.02 Mpg123 0.59r All compiled from source with kernel 2.6.9-1.6 on Fedora Core 2 Any help would be very much appreciated..... The error I am getting is -- Executing WaitMusicOnHold("SIP/snom-james-849d", "30") in new stack Dec 12 00:27:29

Charles S. Antrim wrote: > I am using a card that has an fxo and fxs module. I am no where near an expert but I have my sip phone working through my pstn line and this is my config. /etc/asterisk/sip.conf [general] port = 5060 bindaddr = 192.168.69.1 context = sip disallow = gsm allow = alaw disallow = ulaw nat=disable srvlookup=no localnet=192.168.69.0/255.255.255.0 subscribecontext =

If you set "dialout=fromvm" in your voicemail.conf, how do you then go about being able to dial back out? Is there a service feature code?

I've been reading the Linejack strikes again messages, and have another Newbie question is it possible to use a Voip Product as a Dialout line for * ? I have a Vegastream 100 Voip to PRI. box. With * can I use that as a Dialout / dialin box? The Vega100 does either sip or h.323. Thanks. Bradley Greep

Hi, I am having some trouble with ISDN Dialout. Using a Netjet-s PCI Card. When in Minicom, the only way I can dialout is if i issue ATS18=1 First. Otherwise I get a BUSY message. So thats fine. But when I dialout from asterisk, I get an immediate hangup, so my guess is that asterisk is not issuing ATS18=1 to the ttyI device. Here are my configs, any input would be greatly appriciated.

Hello All. I am trying to use RAGI the ruby agi framework with trixbox. I am having a problem with the dialout part. The RAGI framework creates a file in the /var/spool/asterisk/outgoing directory and routes the call to an extension (I have listed the relevent portion of the file below). The problem is that the initial dial command does not execute properly in trixbox. I am hoping somebody who

I am trying to find a way to allow dialout from voicemail when connected from an 'internal' extension context, but prevent dialout when connected from an 'external' extension context. As far as I can tell the dialout context that can be set in voicemail has no regard for the context from which the call to voicemail came in. Any ideas on this? Maybe a variable passed when

Hi! I have an automated dialout via a call file to a mobile. Can I detect when the call is not answered but forwarded to the mobile operator voicebox? I would like to stop the dialout if this is the case. TIA, Mike

Hi, When I try to dialout over chan_capi everything works fine when I settle for msn=* in my capi.conf and use the primary msn of my ISDN-line. But trying to configure a different MSN the chan_capi doesn't dial and comes with: No one is available to answer at this time What can be the prob? -- Thanks, Marc aka IzNoGood

Dear all I read on the list back in 2003 that * does not support IXJ LineJACK dialout yet is this still the case? Thanks

Hello my name is Ilija Poznic and I have a problem. My configuration is 1. Digium TDM4000P with one FXS. 2. AVM Fritz ISDN adapter (configured with capi). When I connect to my ISP and then start *. Asterisks is registering me to SIP provider iconnect. After that I can call international call trough VoIP. My problem is that I want to dialout to ISP only when I have a international call.

I previously posted about this problem and received suggestions involving turning off echo cancellation. As far as I can tell, echo cancellation is already disabled on this channel, so I'm back. What I've got is a small home setup with a single four-port Digium card: Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXS/DPO Module 3: